IP networks generally provide an excellent infrastructure for geographically distributing components of a telecommunication system. The underlying IP network is optimal for transmission for control signaling, and, when bandwidth is available, can provide an acceptable Quality of Service (or QoS) or Grade of Service (or GOS) for voice communications. When insufficient network resources are available for voice communications or one or more IP network components are down, voice communications can be adversely impacted.
A number of techniques have been attempted to address these issues.
In one technique, if a system had multiple communication gateways controlled by a single controller and the private switching facilities inter-connecting these gateways failed, users can “dial-out” on a public network trunk using the public address (or Direct Inward Dialing or DID number) of the destination party. This approach requires manual intervention by the user first to recognize that a problem exists, second to determine how to circumvent it, and third to dial the DID number. Normally, the calling party would dial only an extension to reach the destination party. If the destination party to be reached does not have a public number, he or she is not reachable by the alternate network.
In another technique known as PSTN Fallback™ of Avaya Inc., a call is forced to the PSTN when an IP trunk connection experiences an unacceptable QoS or GOS. With reference to FIG. 1, a multi-enterprise architecture is depicted, each enterprise 100 and 104 having a separate, independent, and active or primary media servers 112 and 116 with resident call controller functionality. Each enterprise also includes a plurality of digital stations 120 and 124, a plurality of IP or Internet Protocol stations 128 and 132, a gateway 136 and 140 and a Local Area Network or LAN 144 and 148. The media servers 112 and 116 are independent in that one media server in one enterprise is generally unaware of the subscriber configuration information, such as extensions, of the other enterprise's subscribers. The gateways 136 and 140 are interconnected by the Public Switched Telephone Network or PSTN 148 and Wide Area Network or WAN 152. When a call is to be placed over the WAN 152, the originating call controller determines the currently measured network delay and packet loss. When either measured variable reaches a predetermined threshold, the call controller automatically takes the idle IP trunk ports out-of-service, i.e., it busies out the ports. The ports remain out-of-service until the measurements return to the low threshold. No new calls are allowed over the IP trunk. Normal or conventional call routing over the PSTN 148 is used for access to the next preference in the rout pattern.
In another technique known as Separation of Bearer and Signaling™ (SBS) of Avaya Inc., the signaling channel for a call is routed over the WAN 152 while the bearer channel is routed over the PSTN 148. The signaling channel in SBS includes SBS call-control signaling and QSIG private-networking protocol information. SBS associates the signaling and bearer channels at the SBS originating and terminating nodes so that they appear to the end users to be a normal, non-separated call. The use of the WAN for signaling traffic and the PSTN for voice bearer traffic addresses a customer need for using small amounts of bandwidth in the IP WAN for signaling traffic, with the voice bearer portion of the call being sent over inexpensive PSTN facilities. Like PSTN Fallback, SBS™ is used in multi-enterprise calls with each enterprise having separate, independent, and active media servers.
PSTN Fallback™ and SBS™ address architectures where there exist multiple, separate system implementations inter-connected by a traditional inter-switch trunking protocol; in other words, they permit inter-connection only of peer-to-peer systems. With the move to larger, single-server IP WAN-connected media gateway distributed systems, there is no longer a need for IP trunks and SBS. Using trunk group administration to limit bandwidth between media servers is not required nor is PSTN Fallback™ when the number of calls exceeds the administered IP trunk member limit. There is no need to embed an intelligent signaling interface between servers over IP WAN facilities given that the system has only a single active or primary server and that all calls across the system appear to be station-to-station calls.
Another technique for managing IP bandwidth usage includes call admission control in which the number of calls across the WAN or the bandwidth available for voice calls is limited. Call admission control can result in the call being denied and being forwarded to the callee's voice mail server (if accessible), thereby causing caller frustration.
There is a need, particularly in a single-server system, for a call control system that manages IP bandwidth usage effectively, particularly during high traffic periods and/or provides an alternate communication path in the event of problems with the WAN.